1. Field of the Invention
The present invention relates to a test tone determination method and a sound field correction apparatus using the test tone determination method.
2. Description of the Related Art
Due to prevalence of digital versatile discs (DVD) and digital broadcasting, multichannel audio systems, such as home theater systems, are becoming widely used in homes. In such situations, there is an increased demand for users to perform setting of each channel and setting between channels, such as setting of volume balance and frequency characteristics, in multichannel audio systems.
However, since setting and adjustment in multichannel audio systems are complicated, listeners (or users) who are not familiar with such operations may feel puzzled. Thus, in order to simplify or eliminate the necessity for setting and adjustment by listeners, there is a trend in which, when audio playback is performed, an apparatus, such as an AV (audio and visual) amplifier, constituting a multichannel audio system performs correction processing.
Such correction processing is called “automatic sound field correction” or the like. In such correction processing, acoustic conditions of a playback sound field are automatically measured and analyzed, and sound field correction is performed in accordance with an analysis result. That is, in general, as shown in FIG. 4A, the correction processing described below is performed.
(A) A predetermined test tone is output from a speaker SP for a certain channel. An impulse signal, a time stretched pulse (TSP) signal, or a burst wave signal is used as a test tone.
(B) The test tone mentioned in (A) is picked up by a microphone M0 set at the listening position of a listener.
(C) A rising point of an output signal of the microphone M0 is analyzed, and a distance from the speaker SP to the microphone M0 is calculated.
(D) Processing in (A) to (C) is performed for other channels.
(E) An audio signal is processed such that a constant delay time can be achieved between speakers for the individual channels to the listening position (microphone M0) in accordance with results acquired by the processing (D).
In addition, as shown in FIG. 4A, a method for setting microphones M1 and M2, which serve as sound pickup microphones, at the listening positions of a listener and for calculating the distance and angle (direction) between the speaker SP and each of the microphones M1 and M2 using triangulation is also available.
Known technologies are described, for example, in Japanese Unexamined Patent Application Publication No. 2000-261900 and Japanese Patent Application No. 2005-141615 (specification and drawings).
When the distance from the speaker SP to the microphone M0 is measured, variation, peaks, dips, and the like in the frequency characteristics in a playback sound field may affect a measurement result.
In that respect, when the distance from the speaker SP to each of the microphones M1 and M2 is measured, variation, peaks, dips, and the like in the frequency characteristics in a playback sound field may be flexibly coped with. Thus, more appropriate sound field correction can be achieved. It is desirable that sound field correction be performed by calculating the distance or angle between the speaker SP and each of the microphones M1 and M2.
However, in the situation shown in FIG. 4B, measurement using the microphones M1 and M2 is not performed successfully. That is, in the situation shown in FIG. 4B, in order to keep a predetermined distance between the microphones M1 and M2, the microphones M1 and M2 are fixed to an arm or the like. In addition, it is assumed that reflectors are located near the microphones M1 and M2 and that an obstacle is located on a virtual line connecting the speaker SP and the microphone M2. Here, furniture, a wall, a ceiling, or the like corresponds to each of the reflectors, and the body of a listener or a family member, furniture, or the like corresponds to the obstacle.
When acoustic waves of a test tone are output from the speaker SP, an acoustic wave W1 directly reaches the microphone M1, and an acoustic wave WQ1 is reflected by one of the reflectors and then reaches the microphone M1. In addition, an acoustic wave W2 is diffracted and attenuated by the obstacle and directly reaches the microphone M2, and an acoustic wave WQ2 is reflected by the other reflector and then reaches the microphone M2. That is, the acoustic waves W1 and W2 are direct waves, and the acoustic waves WQ1 and WQ2 are indirect waves (reflected waves). In this case, due to attenuation, the amplitude of the direct wave W2 is smaller than that of the indirect wave WQ2. In addition, the indirect wave WQ2 is delayed compared with the direct wave W2.
Thus, output signals SM1 and SM2 of the microphones M1 and M2 in this case are as shown in FIG. 5A. That is, FIGS. 5A, 5B, and 5C schematically show envelopes of the output signals SM1 and SM2 of the microphones M1 and M2 when an impulse signal is supplied as a test tone signal to the speaker SP.
In the playback environment shown in FIG. 4B, as the output signal SM1 of the microphone M1, pulse amplitude P1 acquired by picking up the direct wave W1 is obtained, and then, pulse amplitude Q1 acquired by picking up the indirect wave WQ1 is obtained, as shown in FIG. 5A. In addition, as the output signal SM2 of the microphone M2, pulse amplitude P2, which is small, acquired by picking up the indirect wave W2 is obtained, and then, pulse amplitude Q2 acquired by picking up the indirect wave WQ2 is obtained.
FIGS. 6A and 6B show the main portions of wave shapes of the output signals SM1 and SM2 of the microphones M1 and M2 that are actually observed. In FIGS. 6A and 6B, the horizontal axis represents sample numbers when the output signals SM1 and SM2 are sampled at a frequency of 48 kHz. Thus, the horizontal axis also serves as a time axis. Here, a test tone is an impulse signal, and the point in time when the impulse signal is generated serves as the starting point (origin) of the horizontal axis.
As is clear from FIGS. 6A and 6B, in the environment shown in FIG. 4B, the output signal SM1 includes the large amplitude P1 corresponding to the direct wave W1 and the slightly smaller amplitude Q1 corresponding to the indirect wave WQ1. In addition, the output signal SM2 includes the small amplitude P2 corresponding to the attenuated direct wave W2 and the large amplitude Q2 corresponding to the indirect wave WQ2. The amplitude P2 is almost buried in noise.
In the state shown in FIG. 5A (and FIGS. 6A and 6B), when the presence or absence of the amplitudes P1 and P2 is determined on the basis of, for example, a threshold level VTH, the presence of the amplitude Q2 is detected, instead of the amplitude P2. The distance and angle between each of the microphones M1 and M2 and the speaker SP should be calculated on the basis of the impulse signal supplied to the speaker SP and the rising time of each of the amplitudes Pi and P2. However, since the amplitude Q2 is erroneously determined to be the amplitude P2 in the case shown in FIG. 5A, the distance and angle between each of the microphones M1 and M2 and the speaker SP are calculated on the basis of the rising time of the amplitude Q2, instead of the rising time of the amplitude P2. Thus, an error occurs in calculation of the distance and angle.
If the distance from the speaker SP to the microphone M0 is measured as shown in FIG. 4A, calculating the distance on the basis of the rising time of an indirect wave corresponding to the indirect wave WQ1 or WQ2 is not a significant problem. This is because the indirect wave has an energy larger than that of the direct wave at the position of the microphone M0 that picks up the indirect wave. Thus, in terms of sound field correction, a path in which the indirect wave is reflected can also be included for calculation of distance.
However, when distance is calculated with respect to each of the microphones M1 and M2 or with respect to each of a larger number of microphones, erroneous determination of the amplitude Q2 and the amplitude P2 is performed as long as output signals of the individual microphones are analyzed independently. Thus, incorrect distance and angle are calculated from the amplitude P1 and the amplitude Q2.
In addition, in cases other than the case shown in FIG. 5A, mismatching of output signals of the microphones M1 and M2 may occur. For example, when a loud noise is unexpectedly picked up by one of the microphones M1 and M2, a noise signal may be erroneously determined to be an amplitude corresponding to a direct wave.
Alternatively, when sound field correction is performed by analyzing output signals of the microphones M1 and M2, if distance and angle are calculated by using part of the analysis processing, pre-echo may occur before an amplitude corresponding to a direct wave. This pre-echo may be erroneously determined to be an amplitude corresponding to the direct wave. That is, when a TSP signal is used as a test tone, inverse TSP processing is performed in the process of analysis processing to acquire an impulse response. However, if the spatial impulse response does not sufficiently converge with respect to the length of the TSP signal, due to cyclic convolution of frequency transformation (FFT (Fast Fourier Transform)/IFFT (Inverse Fast Fourier Transform)), a false large amplitude (pre-echo) may appear before the amplitude corresponding to the direct wave. The pre-echo may be erroneously determined to be the amplitude corresponding to the direct wave.
FIG. 5B shows an example of the output signals SM1 and SM2 in such a case. In FIG. 5B, a case where, due to noise or pre-echo, amplitude R2 that exceeds the threshold level VTH appears before the amplitude P2 is shown. In this case, the amplitude R2 is erroneously determined to be the amplitude P2. Thus, incorrect distance and angle are calculated from the amplitude P1 and the amplitude R2.
In addition, for example, as shown in FIG. 5C, when the amplitudes P1 and P2 corresponding to the direct waves W1 and W2 are determined, the time difference T12 between the determined amplitudes P1 and P2 may be too large. That is, when “d” represents the distance between the microphones M1 and M2 and “Td” represents the time period necessary for transmission of an acoustic wave over the distance d, that is, the temporal distance between the microphones M1 and M2 with respect to an acoustic wave, the time difference T12 between the amplitudes P1 and P2 reaches maximum when the speaker SP and the microphones M1 and M2 are disposed in a line, and the time difference T12 should not be larger than the time period Td. However, in some cases, such as due to the occurrence of a system error or the presence of an extremely large obstacle, the time difference T12 may be larger than the time period Td.
In a case where erroneous determination of an amplitude corresponding to a direct wave is performed as described above, when output signals of the microphones M1 and M2 are analyzed, a triangle connecting the speaker SP and the microphones M1 and M2 is not formed. Thus, the distance and angle between the speaker SP and each of the microphones M1 and M2 is not calculated correctly.